Q-AIMD: A Congestion Aware Video Quality Control Mechanism

Following the constant increase of the multimedia traffic, it seems necessary to allow transport protocols to be aware of the video quality of the transmitted flows rather than the throughput. This paper proposes a novel transport mechanism adapted to video flows. Our proposal, called Q-AIMD for video quality AIMD (Additive Increase Multiplicative Decrease), enables fairness in video quality while transmitting multiple video flows. Targeting video quality fairness allows improving the overall video quality for all transmitted flows, especially when the transmitted videos provide various types of content with different spatial resolutions. In addition, Q-AIMD mitigates the occurrence of network congestion events, and dissolves the congestion whenever it occurs by decreasing the video quality and hence the bitrate. Using different video quality metrics, Q-AIMD is evaluated with different video contents and spatial resolutions. Simulation results show that Q-AIMD allows an improved overall video quality among the multiple transmitted video flows compared to a throughput-based congestion control by decreasing significantly the quality discrepancy between them

Application-Level QoS: Improving video conferencing quality through sending the best packet next

In a traditional network stack, data from an application is transmitted in the order that it is received. An algorithm is proposed where information about the priority of packets and expiry times is used by the transport layer to reorder or discard packets at the time of transmission to optimise the use of available bandwidth. This can be used for video conferencing to prioritise important data. This scheme is implemented and compared to unmodified datagram congestion control protocol (DCCP). This algorithm is implemented as an interface to DCCP and tested using traffic modelled on video conferencing software. The results show improvement can be made to video conferencing during periods of congestion - substantially more audio packets arrive on time with the algorithm, which leads to higher quality video conferencing. In many cases video packet arrival rate also increases and adopting the algorithm gives improvements to video conferencing that are better than using unmodified queuing for DCCP. The algorithm proposed is implemented on the server only, so benefits can be obtained on the client without changes being required to the client

Towards a versatile transport protocol

n the context of a reconfigurable transport protocol, this paper introduces two protocol instances based on the com- position and specialisation of the TFRC congestion control and Selective Acknowledgment mechanisms. The two result- ing transport architectures lead respectively to the QTP_AF protocol, specifically designed to operate over QoS-enabled networks and the QTP_light protocol, specifically designed for resource-limited end systems connected to powerful servers. QTP_AF combines QoS-aware TFRC congestion control with full reliability to provide a transport service similar to TCP but additionally taking into account network-level band-width reservations. QTP_light proposes a modification of TFRC that shifts from the receiver to the sender the complexity of the loss rate estimation mechanism. This modification allows to alleviate the processing and communication load of "light" resource limited mobile receivers. We present the concept of these protocols and their adaptation in the EuQoS European project framework

Rate Control State-of-the-art Survey

The majority of Internet traffic use Transmission Control Protocol (TCP) as the transport level protocol. It provides a reliable ordered byte stream for the applications. However, applications such as live video streaming place an emphasis on timeliness over reliability. Also a smooth sending rate can be desirable over sharp changes in the sending rate. For these applications TCP is not necessarily suitable. Rate control attempts to address the demands of these applications. An important design feature in all rate control mechanisms is TCP friendliness. We should not negatively impact TCP performance since it is still the dominant protocol. Rate Control mechanisms are classified into two different mechanisms: window-based mechanisms and rate-based mechanisms. Window-based mechanisms increase their sending rate after a successful transfer of a window of packets similar to TCP. They typically decrease their sending rate sharply after a packet loss. Rate-based solutions control their sending rate in some other way. A large subset of rate-based solutions are called equation-based solutions. Equation-based solutions have a control equation which provides an allowed sending rate. Typically these rate-based solutions react slower to both packet losses and increases in available bandwidth making their sending rate smoother than that of window-based solutions. This report contains a survey of rate control mechanisms and a discussion of their relative strengths and weaknesses. A section is dedicated to a discussion on the enhancements in wireless environments. Another topic in the report is bandwidth estimation. Bandwidth estimation is divided into capacity estimation and available bandwidth estimation. We describe techniques that enable the calculation of a fair sending rate that can be used to create novel rate control mechanisms.Peer reviewe

Design and analysis for TCP-friendly window-based congestion control

The current congestion control mechanisms for the Internet date back to the early 1980’s and were primarily designed to stop congestion collapse with the typical traffic of that era. In recent years the amount of traffic generated by real-time multimedia applications has substantially increased, and the existing congestion control often does not opt to those types of applications. By this reason, the Internet can be fall into a uncontrolled system such that the overall throughput oscillates too much by a single flow which in turn can lead a poor application performance. Apart from the network level concerns, those types of applications greatly care of end-to-end delay and smoother throughput in which the conventional congestion control schemes do not suit. In this research, we will investigate improving the state of congestion control for real-time and interactive multimedia applications. The focus of this work is to provide fairness among applications using different types of congestion control mechanisms to get a better link utilization, and to achieve smoother and predictable throughput with suitable end-to-end packet delay

Towards a sender-based TCP friendly rate control (TFRC) protocol

Pervasive communications are increasingly sent over mobile devices and personal digital assistants. This trend is currently observed by mobile phone service providers which have measured a significant increase in multimedia traffic. To better carry multimedia traffic, the IETF standardized a new TCP Friendly Rate Control (TFRC) protocol. However, the current receiver-based TFRC design is not well suited to resource limited end systems. In this paper, we propose a scheme to shift resource allocation and computation to the sender. This sender-based approach led us to develop a new algorithm for loss notification and loss-rate computation. We detail the complete implementation of a user-level prototype and demonstrate the gain obtained in terms of memory requirements and CPU processing compared to the current design. We also evaluate the performance obtained in terms of throughput smoothness and fairness with TCP and we note this shifting solves security issues raised by classical TFRC implementations

Transport Layer Optimizations for Heterogeneous Wireless Multimedia Networks

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.Ph.D.Committee Chair: Vijay Madisetti; Committee Member: Raghupathy Sivakumar; Committee Member: Sudhakar Yalamanchili; Committee Member: Umakishore Ramachandran; Committee Member: Yucel Altunbasa